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/*
* Pulseaudio input
* Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* PulseAudio input using the simple API.
* @author Luca Barbato <lu_zero@gentoo.org>
*
*/
#include <pulse/simple.h>
#include <pulse/rtclock.h>
#include <pulse/error.h>
#include "libavformat/avformat.h"
#include "libavformat/internal.h"
#include "libavutil/opt.h"
#define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE)
typedef struct PulseData {
AVClass *class;
char *server;
char *name;
char *stream_name;
int sample_rate;
int channels;
int frame_size;
int fragment_size;
pa_simple *s;
int64_t pts;
int64_t frame_duration;
} PulseData;
static pa_sample_format_t codec_id_to_pulse_format(int codec_id) {
switch (codec_id) {
case CODEC_ID_PCM_U8: return PA_SAMPLE_U8;
case CODEC_ID_PCM_ALAW: return PA_SAMPLE_ALAW;
case CODEC_ID_PCM_MULAW: return PA_SAMPLE_ULAW;
case CODEC_ID_PCM_S16LE: return PA_SAMPLE_S16LE;
case CODEC_ID_PCM_S16BE: return PA_SAMPLE_S16BE;
case CODEC_ID_PCM_F32LE: return PA_SAMPLE_FLOAT32LE;
case CODEC_ID_PCM_F32BE: return PA_SAMPLE_FLOAT32BE;
case CODEC_ID_PCM_S32LE: return PA_SAMPLE_S32LE;
case CODEC_ID_PCM_S32BE: return PA_SAMPLE_S32BE;
case CODEC_ID_PCM_S24LE: return PA_SAMPLE_S24LE;
case CODEC_ID_PCM_S24BE: return PA_SAMPLE_S24BE;
default: return PA_SAMPLE_INVALID;
}
}
static av_cold int pulse_read_header(AVFormatContext *s,
AVFormatParameters *ap)
{
PulseData *pd = s->priv_data;
AVStream *st;
char *device = NULL;
int ret;
enum CodecID codec_id =
s->audio_codec_id == CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
const pa_sample_spec ss = { codec_id_to_pulse_format(codec_id),
pd->sample_rate,
pd->channels };
pa_buffer_attr attr = { -1 };
st = avformat_new_stream(s, NULL);
if (!st) {
av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
return AVERROR(ENOMEM);
}
attr.fragsize = pd->fragment_size;
if (strcmp(s->filename, "default"))
device = s->filename;
pd->s = pa_simple_new(pd->server, pd->name,
PA_STREAM_RECORD,
device, pd->stream_name, &ss,
NULL, &attr, &ret);
if (!pd->s) {
av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
pa_strerror(ret));
return AVERROR(EIO);
}
/* take real parameters */
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = codec_id;
st->codec->sample_rate = pd->sample_rate;
st->codec->channels = pd->channels;
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
pd->pts = AV_NOPTS_VALUE;
pd->frame_duration = (pd->frame_size * 1000000LL * 8) /
(pd->sample_rate * pd->channels * av_get_bits_per_sample(codec_id));
return 0;
}
static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
{
PulseData *pd = s->priv_data;
int res;
pa_usec_t latency;
if (av_new_packet(pkt, pd->frame_size) < 0) {
return AVERROR(ENOMEM);
}
if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) {
av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n",
pa_strerror(res));
av_free_packet(pkt);
return AVERROR(EIO);
}
if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) {
av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n",
pa_strerror(res));
return AVERROR(EIO);
}
if (pd->pts == AV_NOPTS_VALUE) {
pd->pts = -latency;
}
pkt->pts = pd->pts;
pd->pts += pd->frame_duration;
return 0;
}
static av_cold int pulse_close(AVFormatContext *s)
{
PulseData *pd = s->priv_data;
pa_simple_free(pd->s);
return 0;
}
#define OFFSET(a) offsetof(PulseData, a)
#define D AV_OPT_FLAG_DECODING_PARAM
static const AVOption options[] = {
{ "server", "pulse server name", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
{ "name", "application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D },
{ "stream_name", "stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
{ "sample_rate", "sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, D },
{ "channels", "number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, D },
{ "frame_size", "number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.dbl = 1024}, 1, INT_MAX, D },
{ "fragment_size", "buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.dbl = -1}, -1, INT_MAX, D },
{ NULL },
};
static const AVClass pulse_demuxer_class = {
.class_name = "Pulse demuxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_pulse_demuxer = {
.name = "pulse",
.long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"),
.priv_data_size = sizeof(PulseData),
.read_header = pulse_read_header,
.read_packet = pulse_read_packet,
.read_close = pulse_close,
.flags = AVFMT_NOFILE,
.priv_class = &pulse_demuxer_class,
};
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